1. Field of the Invention
This invention relates to a digital audio mixer which mixes a plurality of digital audio signals gathered by a large number of microphones or like means to produce a single complete audio program, and more particularly to a digital audio signal mixer suitable editing an audio signal provided from a digital video tape recorder.
2. Description of the Related Art
It is a popular practice, in order to mix audio signals from a plurality of systems, to produce an audio program by means of a digital audio signal mixer (hereinafter referred to simply as digital mixer) irrespective of whether the signal sources are analog signal sources or digital signal sources.
An exemplary digital mixer wherein digital audio signals in a plurality of channels are mixed at a desired mixture ratio to obtain new digital audio signals in a second plurality of channels is constructed as shown in FIG. 1.
Referring to FIG. 1, the digital mixer shown therein is constructed such that it receives digital audio signals in 32 channels and outputs a program of digital audio signals in each of 4 channels. The digital audio signals #1 to #32 corresponding to 32 channels are received at input terminals 1 of the digital mixer and inputted to a routing switcher 2. The routing switcher 2 selects audio signals corresponding to 16 channels (CH1 to CH16) from the 32 channels of input audio signals. Each of the 16 output channels of the routing switcher 2 is processed successively by a mute switch 3, a de-emphasis and phase variation unit 4, a delay unit 5, a filter 6, an equalizer 7 and an insertion circuit 8, and finally, the sound volume for the channel is adjusted by a channel fader 9. Thereafter, the 16 channel signals are added by way of assignment switches 10 on four buses and then adjusted in sound volume by master faders 11 to make four programmed outputs 12 (PGM1 to PGM4).
More particularly, the digital audio signals #1 to #32 corresponding to the 32 channels are digital audio data and normally are signals in the AES/EBU format. In particular, the digital audio signals may be digital output signals from digital audio tape recorders or compact disk players and audio PCM (pulse code modulation) data signals to which additional information is added.
The routing switcher 2 determines which of the digital audio signals #1 to #32 should be allotted to the channels CH1 to CH16, and can allot the same signal, for example, the input signal #1, to all of the channels CH1 to CH16. The mute switches 3 are used to change over the signals in the channels CH1 to CH16 so that any digital audio signal which is not required by the operator may not be transmitted.
Each of the de-emphasis and phase variation units 4 includes a de-emphasis circuit which reverses emphasis processing with respect to input signals of which the high frequency band has been subjected to pre-emphasis.
Each of the de-emphasis and phase variation units 4 further includes a phase variation circuit which reverses the phase of a digital audio signal. Since such phase variation circuit is not required for playing back a compact disk, a music tape or a record the phase variation function is not normally provided for an audio amplifier provided in consumer audio equipment.
The phase variation circuit is used principally to provide phase correction of a signal produced by a microphone during sound recording. Microphones are constructed with a diaphragm that is vibrated by sound pressure and the vibrations of the diaphragm are converted into an electric signal. Microphones are divided into two types including a first type wherein a positive voltage is generated at non-dense portions of acoustic vibrations and a second type wherein a positive voltage is generated at dense portions of acoustic vibrations. Accordingly, when sound recording is performed using a plurality of microphones, the phases of the output signals from the microphones must be matched since the signals may have different polarities. If the phases are not matched, the acoustic vibrations of the microphones may cancel each other so that a low sound which is particularly low in directivity may fade away. Therefore, input signals provided for mixing by microphones must necessarily be matched in phase depending upon characteristics of the microphones, and phase reversers for this purpose are required for a mixer.
When an audio signal output from a digital video tape recorder is to be edited, and editing of the video image is performed by means of a digital multi-effector (DME) or a like apparatus, then the video image may be delayed by several frames. In such a case, the sound must also be delayed by an equal amount of time, and the delay units 5 are used to effect such a delay with respect to sound.
Each of the filters 6 includes a low frequency band cut filter and a high frequency band cut filter and is used, in the mixer, to remove artifacts and noise. The high frequency band cut filter is used to remove, for example, hysteresis noise which may be produced upon reproduction of a tape or a like recording medium on which an analog audio signal is recorded. Meanwhile, the high frequency band cut filter is used to remove surrounding low frequency noise such as a sound that resembles wind.
Each of the equalizers 7 is used to raise or lower the signal level in a certain sound region of an audio signal and is employed, in the mixer, as an effector device principally for making sound effects.
Each of the insertion circuits 8 has the function of releasing an external contact when the operator wishes to insert an external effector device (a limiter, a filter, an equalizer or the like) into the mixer. In particular, the insertion circuit 8 cuts the audio signal path at an insertion point and connects the cut path to the outside. Accordingly, if an insertion function is rendered operative but no connection is made to an external device, the sound is interrupted at the insertion point, and the input signal will therefore not be outputted.
Referring particularly to FIG. 31, when the insertion function is to be rendered inoperative, a switch 13 is switched to the upper connection shown in FIG. 31 so that the input and output are connected to each other, and the insertion function is not exhibited. On the other hand, if the insertion function is to be rendered operative, the switch 13 is switched to the lower connection so that both the input and the output are released to the outside at an insertion point 25. In this case, the operator will connect to the insertion out terminal the input of an effector device which the operator wants to use and will connect the output of the effector device to the insertion in terminal. In this manner, an effect the digital mixer does not have can be applied by way of an external apparatus.
Referring back to FIG. 1, the channel faders 9 have the function of adjusting the sound volumes in channels CH1 to CH16, and volume adjustment (from +12 dB to -.infin.) by the channel faders 9 the resulting signals are sent into the mixing buses by the assignment switches 10 for mixing of the signals in the channels CH1 to CH16.
The assignment switches 10 are each used to determine whether or not the volume-adjusted sound signal should be mixed, and when the switch is on, the sound is added, but when the switch is off, the sound is not added. Thus, each assignment switch 10 is an on/off switch for an input to the mixing buses.
The sound (for example, surround programs) added by the four mixing buses by way of the assignment switches 10 thus forms programs PGM1, PGM2, PGM3 and PGM4, and the outputs signals corresponding to the programs PGM1, PGM2, PGM3 and PGM4 are subject to final sound volume adjusted by the respective master faders 11. (While the range of the adjustment is normally from 0 dB to -.infin., a positive gain may also be provided). After sound volume adjustment has been performed by the master faders 11, the signals are modulated back into signals in the AES/EBU format and outputted as the outputs PGM1 to PGM4 to the outside.
While the principal parts of the digital mixer for digital audio signals have been described above, problems in the conventional digital mixer will be described below. First, problems involved in conventional phase variation sound volume adjustment will be described.
Conventionally, a digital mixer having a phase variation function realizes phase reversal either by multiplying audio data by "1" by means of a multiplier 14 or by multiplying audio data by "-1" by means of another multiplier 15 as shown in FIG. 28(A). With the present method, however, even if a signal 28A which is not reversed in phase is continuous as shown in FIG. 28(B), when a phase reversal switch 16 is changed over to reverse the phase of the signal 28A, a discontinuous waveform 28B may be obtained depending upon the timing at which such phase reversal takes place, and as a result, a phase conversion waveform such as that shown in FIG. 28(C) may be obtained. Since a point of discontinuity is produced, noise is produced by the change over.
Particularly in the case of a digital audio signal, the signal has the form of discrete data, so that a phase conversion waveform as shown in FIG. 28(C) results in loss of data continuity, and consequently, a unique noise is produced. The presence of a discontinuous point in the digital audio signal produces frequency components that extend up to infinity, and when the signal is actually converted from the digital form into the analog form, folded noise is produced because frequency components higher than one-half of the sampling frequency cannot be represented. Thus, when a digital audio signal is to be changed over, noise will be produced unless the change over can be performed so as not to produce a discontinuous point.
In order to actually apply a phase variation function to a digital audio signal, it is a conventional practice to perform multiplication by "-1" using a DSP (digital signal processor) as shown in FIG. 28(A) to achieve variation in phase. When multiplication is performed using a DSP, the multiplication coefficient k normally assumes a value of the range -1.ltoreq.k.ltoreq.1, and processing is performed so that, when an input signal is multiplied by "1", the input signal itself is obtained; when the input signal is multiplied by "0.5" a signal having a level one half that of the input signal is obtained; when the input signal is multiplied by "0", no sound is obtained; when the input signal is multiplied by "-1", a signal having a phase opposite to that of the input signal is obtained; and when the input signal is multiplied by "-0.5" a signal having a level one half that of the input signal and having a phase opposite to that of the input signal is obtained. Accordingly, when it is desired to obtain a phase reversing function for a digital audio signal, the function can be realized with a circuit for selecting whether an input signal is to be multiplied by "1" or "-1". However, such a phase variation technique does not provide a solution to the problem of signal discontinuity, and the waveform shown in FIG. 28 will be produced.
Problems related to a fader which performs sound volume adjustment in the digital mixer will be now described.
When a digital audio signal is digitally adjusted in sound volume by means of a fader or a similar device, the actual signal processing is performed using a DSP. To carry out the processing, the fader multiplies the digital audio signal by a coefficient having a value ranging from 1 to 0. For example, if the digital audio signal is multiplied by "1", the input signal maintains its level but if it multiplied by "0" the level of the input signal is reduced to zero, that is, no sound. FIG. 29 shows an example of a conventional digital fader.
Referring to FIG. 29, a control value output from a fader 17 is converted into digital data by an analog to digital converter 18 and is read and converted into coefficient data ranging from 1 to 0 by a CPU (central processing unit) 19. The coefficient data is written into a data interpolator 20.
In actual operation of the circuit shown in FIG. 29, the timings of the coefficient value transferred from the CPU 19 to which the coefficient value is applied are shown in FIG. 6(C). Referring to FIG. 6(C), the digital audio data is updated with a period of 1/fs (where fs is a sampling frequency). Meanwhile, since the digital audio signal is communicated in the form of a serial signal inside of the mixer, the varying digital audio data values No. 1, No. 2, . . . , No. N are provided as seen in FIG. 6(C). On the other hand, when fader coefficients Fdn-1, . . . are to be written, it is almost impossible for the CPU 19 to vary the coefficient for each 1/fs period in synchronism with the sampling frequency fs of the digital audio signal.
Two problems are involved: (1) While a master clock signal of the CPU 19 must be synchronized with the sampling frequency fs of the digital audio signal, an apparatus having a plurality of inputs and outputs such as a mixer cannot determine with which one of the plural inputs the apparatus should be synchronized or how to proceed when an out-of-synchronization condition occurs, and accordingly, it is difficult to assure proper operation of the CPU.
(2) Even if the problem described above in paragraph (1) could be overcome, so that the CPU could re-write the coefficient in synchronism with the sampling frequency fs of the digital audio signal, there would be a very heavy processing burden on the CPU to re-write the coefficient for each 1/fs period. That is it might be necessary to provide one CPU for each fader, which is not practical.
Accordingly, the fader coefficient Fdn-1, Fdn or Fdn+1 can be varied only once over several variations of the digital audio signal as seen from FIG. 6(C). Actually, in conventional digital mixers, the sampling frequency fs of the audio signal is 44.1 kHz or 48 kHz while the period for updating a fader coefficient is 60 Hz, and accordingly, the fader coefficient is varied over about 1,000 variations of the audio data.
The fader coefficient will vary in a stepwise fashion as shown by the "real data" in FIG. 6(B). If the coefficient is used as it is for multiplication of the digital audio data by a multiplier 21 that is part of the DSP shown in FIG. 29, then modulation noise is formed at the transfer period (60 Hz) of the coefficient and will produce a rumbling noise (known as "dipper noise").
Therefore, in the DSP of FIG. 29, the fader coefficient transferred from the CPU 19 is not used as it is but rather is used for multiplication after interpolation by the interpolator 20. Thus, use of interpolation data as illustrated in FIG. 6(B) will be described.
If a non-interpolated coefficient which exhibits a stepwise variation as shown in FIG. 6(B) is used, then modulation noise is produced as described hereinabove. Therefore, in order to interpolate the coefficient within the DSP, two different time constants of 20 ms and 5 ms are employed as the time constant for interpolation. Determination of the appropriate time constants is based on results of an actual listening test.
The test method will now be described. First, the time constant of the interpolator 20 is set to a high value (for example, 200 ms). Thus, when the fader 17 is varied up and down, the time constant is sufficiently high that no modulation noise will be produced, but naturally the fader is slow in reaching a desired value as seen from the curve 6B in FIG. 6(A). In this instance, while no noise is produced, a great deal of time is required before the sound becomes loud after the fader 17 is adjusted to increase the volume or conversely before the sound becomes muted after the fader 17 is adjusted to decrease the volume, and consequently, the perceived change in the sound is delayed. In short, as the time constant becomes smaller, the response becomes faster, and if the time constant is reduced to a value near a limit at which no modulation noise is produced, then interpolation which does not produce noise and has a fast response can be realized. The time constant obtained by such testing is 20 ms.
Referring back to FIG. 29, is a fader control data mute switch 22 is switched on or off and the fader control 17 is disconnected or connected, then although no noise is produced, the response is still perceived to be slow. In short, while an acceptably fast response is obtained upon increasing or decreasing adjustment of the fader control 17, the response is perceived to be slow upon switching on or off of the mute switch 22. Thus, with the mute switch 22 as well, the time constant is gradually reduced until an acceptable response is provided. The time constant obtained in this manner is 5 ms.
In this instance, while it appears that actuation of a switch or a like element in one direction does not particularly produce noise when the coefficient is interpolated, nevertheless, if the fader coefficient is varied rapidly from "0" to a certain value or from a certain value to "0" then a point of discontinuity occurs in the audio data. Upon conversion of the digital audio data having the point of discontinuity into analog data, folded noise is produced according to the sampling theorem, and noise is perceived like that described hereinabove in connection with phase variation.
Thus, the problem is that, if the same time constant is used in the interpolator 20 both when the mute switch 22 is switched on or off and when the fader 17 is operated, and if the time constant is adjusted to be appropriate for the mute switch 22, then noise is produced, but on the other hand, if the time constant adjusted to be appropriate for the operation of the fader 17, then the response is slow.
Problems related to a conventional effector device for a digital mixer will be now described. Conventionally, when sound effect is to be inserted by means of an effector (equalizer or filter), switching of the effector 23 into the system is performed by simple on/off operation of a switch 24 as shown in FIG. 30. However, in the case of a digital audio signal, this will result in loss of data continuity, and consequently, switching noise is sometimes produced.
For example, when an equalizer is used as an effector, and assuming a certain sound region (around f.sub.0) is boosted by a significant factor, for example, by +10 dB as in an EQ frequency characteristic shown in FIG. 30(B), then, if the input audio signal contains sound around f.sub.0, simple switching of the switch 24 as in the conventional arrangement results in a difference in output level depending on whether the switch 24 is switched to the input side or to the effector side. This is illustrated in FIG. 11(B). Referring to FIG. 11(B), switching on the switch 24 causes a point of discontinuity in audio data 11C, as seen from a curve 11D, because of an abrupt change in level of the audio signal.
Problems related to the insertion circuits 8 will now be described. Generally, in an audio mixer, the internal level (head room) is set lower than an input level taking into consideration that audio signals from a plurality of inputs are to be added. If an audio signal is outputted from the insertion point 25 to the outside with such interval level that is lower than the input level, then a full-bit input is not provided to an external effector apparatus and an optimum S/N ratio cannot be obtained. Thus, if an audio signal is outputted from the insertion point 25 (FIG. 31) with the level thereof having been returned to the input level, as when the external effector provides some gain (by boosting high- or low-level sounds), then the data will be clipped as shown in FIG. 13(B).
If the internal level is not lowered, and data clipping occurs, then the data cannot be restored, but if the internal level is set low, then with the level being subsequently lowered by the master fader 11, normal data can be outputted as described above. If input signals are added without change, then clipping naturally occurs, and consequently, even if the master fader 11 is operated to reduce the signal level, the clipped sound is only reduced in volume, but the clipping is not remedied. Accordingly, the internal level of the mixer must be kept low.
As described hereinabove, the insertion point 25 (FIG. 31) is provided at a stage next to the equalizer 7 but preceding the channel fader 9 (FIG. 1), and accordingly, the internal level is lower than an input level. This is done not only to prevent clipping caused by addition of audio data, as described above, but also because the mixer involves effects which produce some gain. For example, an equalizer 7 may increase the sound of a certain band by up to 15 dB. Therefore, if the internal level is not low, even with the signal level being lowered by the channel fader 9, clipping by the equalizer 7 cannot be avoided.
Thus, optimum input and output levels of the insertion circuit depend upon the manner in which the operator uses the insertion circuit.
Moreover, in the mixer described above, when inputs supplied by way of the routing switcher 2 are provided as inputs to the mixer, and if the setting of the routing switcher 2 is changed, then, according to conventional practice, the parameter settings of the channels CH1 to CH16 of the mixer must also be changed. Accordingly, upon changing channels, the parameters must be re-set.
In particular, given that conventionally an equalizer 7, a filter 6, a fader 9 and so forth are provided to perform signed processing for each of the 16 channels CH1 to CH16, according to a conventional snapshot automation technique (Japanese Patent Publication Application No. Heisei 2-47125 and so forth), parameters for the equalizers, the filters, the faders and so forth appropriate to the inputs for the channels CH1 to CH16 are instantaneously read out.
For example, in the case of scene setting data shown for example in FIG. 16, parameters for equalizers, filters, faders and so forth are changed instantaneously in accordance with the scene setting data. In short, a snapshot varies the parameter values with regard to input signals provided from the routing switcher 2. Accordingly, there is a problem in that the snapshot data does not reflect the ultimate source (audio input signal #) which is inputted to channel CH1, for example.
An output (PGM) level indicating apparatus for the mixer described above or as otherwise proposed (Japanese Patent Publication Application No. Heisei 2-47125) will now be described.
Referring to FIG. 17, the mixer described above includes a processor rack 26 for processing data, and a control panel 27 for providing a man-machine interface. The control panel 27 includes a meter 28 for indicating a sound volume/sound quality data level thereon, and data for the meter 28 is produced from data corresponding to the external outputs PGM1 to PGM2 (FIG. 1).
A level indication on the meter 28 is not an indication of the PGM meter data per se but rather is an indication in units of dB. Meter data to be displayed on the meter 28 is produced by the processor 26 as seen from the flow chart presented as FIG. 20.
The mixer of FIG. 1 is an audio mixer of the full digital processing type, and accordingly, the meter data also is in the form of digital data. In general the meter 28 is a bar graph meter having 100 segments, for example, as shown in FIGS. 18(A) to 18(C) or 19(A) to 19(C), and the meter data is 8-bit data.
In a conventional meter indication, the reference level is not typically added to the meter data by the control panel 27 and then sent with the meter data to the meter 28 as shown in the flow chart of FIG. 20; rather, the meter data is produced from audio output data by the processor rack 26 and is transmitted to the control panel 27 so that the bar graph meter 28 is lit in accordance with the meter data while an indication of a reference level is performed by changing the color (for example, from green to orange) of those segments 29 that are higher than a reference level, as shown in FIG. 32. Alternatively, a reference level setting knob 30 and another bar graph meter 31 may be used to indicate a reference level.
Such methods, however, are disadvantageous in that a meter indication and a reference level indication are not easily read together, and the latter method complicates the hardware because the additional bar graph indicator is required.
In addition, for the digital mixer as described above, maintenance is required and it is necessary to test the functioning of the circuit blocks in the mixer. If the mixer does not include a self-diagnosing system which checks the functioning of the circuit blocks, particularly the DSPs and so forth which perform the digital processing, then audio signals at all of the outputs must be checked while inputting audio signals to all of the inputs.
Even if the digital mixer includes a self-diagnosing system, if the mixer does not have a reference indicating function for the DSPs or integrated circuits making up the mixer, then measurements must be performed using measuring instruments or the like. In this manner, a conventional self-diagnosing system cannot eliminate the need for cumbersome jigs and/or tools and complicated maintenance operations.